Hi all, hope everything's fine out there.
Basically there's no REINVITE to join caller party and agent party, and the call stays up, mute. I'm trying to figure out why calls coming from an Asterisk through a SIP Trunk, randomly get this error:
13:17:18.932: SipDialog: event CONNECTED_TIMEOUT_ST, t=324581940, s=7, r=5, m=e6dede8 port=[PORT]
13:17:18.932: CID:CUUID>6647aa0e-ffeb-4b35-ad1b-844073c413af:01L0V41B74A87400R5F2E2LAES06KDI4:
13:17:18.932 SIPCONN([CALLER NUMBER]): HandleSipDialogEvent(CONNECTED_TIMEOUT_ST)
13:17:18.932 SIPCONN([CALLER NUMBER]): new transaction
13:17:18.932 SIPCONN([CALLER NUMBER]): Park(1,40)
13:17:18.932 SIPCONN([CALLER NUMBER]): Park::Error
13:17:18.932 SIPCONN([CALLER NUMBER]): SIPCONN(e6ded10,01L0V49B74A87400R5F2E2LAES0HISG1) -Tr(262198752,SipTransactionAcceptCall)
13:17:18.932: Tr(262198752,SipTransactionAcceptCall):failed
13:17:18.932: Sc(262198746):step 0, Tr(262198752,SipTransactionAcceptCall) - failed
13:17:18.932: Tr(262198746,SipScenario):failed
13:17:18.932: SIPCM: transaction Tr(262198746,SipScenario) failed
13:17:18.932: GetRegistration::Unable to resolve number for DN:dummy
13:17:18.932: PI: 00 S[IN]D[[CALLER NUMBER]]C[*D[[CALLER NUMBER]]]P[MSML]
13:17:18.933: SIPPARTY(10051): handle postponed media service operation
13:17:18.933: media service handling postponed
13:17:18.933: previous scenario Tr(262198746,SipScenario) is not cleaned up
13:17:18.933: Sc(262198746):step 0, Tr(262198851,SipTransactionDetachMediaService) - begin
13:17:18.933 SIPCONN(MSML): DetachMediaPeer
13:17:18.933 SIPCONN([CALLER NUMBER]): ClrMediaPeer
13:17:18.933 SIPCONN(MSML): set monitor deaef90, 0
13:17:18.933 SIPCONN(MSML): state e:1,p:3,s:0,c:0,rc:0,m:0
13:17:18.933: SipDialog: ClearCall(phone=0,state=7)
13:17:18.933: SipDialog::Terminate(state=7,reason=0)
13:17:18.933: SIPDLG[18771797]: register TRN[324589627]
13:17:18.933: Sending [0,UDP] 508 bytes to [SERVERNAME]:[PORT] >>>>>
BYE sip:Genesys@[MEDIA SERVER IP]:[PORT] SIP/2.0
From: <sip:[CALLER NUMBER]@[SIP SERVER IP]:[PORT]>;tag=0061AB84-2B39-1483-9000-D95E270AAA77-47872976
To: <sip:MSML@[SIP SERVER IP]:[PORT]>;tag=6202EB9B-0000-D961-9341-07A0917C9258
Call-ID: 0061AB66-2B39-1483-9000-D95E270AAA77-40936834@[SIP SERVER IP]
CSeq: 3 BYE
Content-Length: 0
Via: SIP/2.0/UDP [SIP SERVER IP]:[PORT];branch=z9hG4bK0061AB8E-2B39-1483-9000-D95E270AAA77-128741076
Max-Forwards: 69
Route: <sip:0x1eb43b00@[SERVER IP]:[PORT];lr;[HOSTNAME]=1;idtag=010A5724>
Any clue?
Best,
Gef
